Digital signal processing (DSP) is the mathematical manipulation of an information signal to modify or improve it in some way. It is characterized by the representation of discrete time, discrete frequency, or other discrete domain signals by a sequence of numbers or symbols and the processing of these signals.

The goal of DSP is usually to measure, filter and/or compress continuous real-world analog signals. The first step is usually to convert the signal from an analog to a digital form, by sampling and then digitizing it using an analog-to-digital converter (ADC), which turns the analog signal into a stream of numbers. However, often, the required output signal is another analog output signal, which requires a digital-to-analog converter (DAC). Even if this process is more complex than analog processing and has a discrete value range, the application of computational power to digital signal processing allows for many advantages over analog processing in many applications, such as error detection and correction in transmission as well as data compression.

TI has a centralized training site where DSP designers can access all of TI’s training webcasts, workshops and seminars. It can be found at www.dspvillage.ti.com/trainingpr2. It covers TI DSP, tools, software and applications. Analog training is also included.

3: Where can I get free software for general DSP?

The packages listed below are mostly not oriented for use with a specific DSP processor. See the later sections in the FAQ for software relevant to a particular programmable DSP chip.

Q1.3.1: DSP Packages for MATLAB

Updated 05/06/02

FOR STUDENTS IN THE US AND CANADA: The MATLAB Student Version, available from The MathWorks, is a full-featured version of MATLAB and includes Simulink (with model sizes up to 300 blocks) and the Symbolic Math toolbox. It is available for Windows and Linux. See http://www.mathworks.com/products/studentversion/.

MATLAB user’s group public domain extensions to MATLAB

Description:

The MATLAB Digest is issued at irregular intervals based on the number of questions and software items contributed by users. To subscribe to the newsletter, send mail to subscribe@mathworks.com. To make submissions to the digest, please send to hwilson@ua1vm.ua.edu with a subject: “DIG” and description.

Wavelet Tools

Description:

There is a set of Wavelet Tools available for MATLAB, see Section 2.9 of this FAQ.

Communications Toolbox

Description:

We have developed a “Communications Toolbox” based on the MATLAB code for classroom use. It is used by students taking a 4th year communications course where the emphasis is on digital coding of waveforms and on digital data transmission systems. The MATLAB code that constitutes this toolbox has been in use for over two years.

There are close to 100 “M-files” that implement various functions. Some of them are quite simple and are based on existing MATLAB M-files. But a great many of them has been created from scratch. We also prepared a lab manual (in TEX format) for the 7 simulations which the students perform as the lab component of this course. The topics of these simulations are:

GSM Routines

Q1.3.5: Text to Speech Conversion Software

Updated 1/7/97

Free (but not public domain) text to speech conversion software is available via anonymous ftp from wilma.cs.brown.edu in the pub directory as speak.tar.Z. It will compile and run on a SPARC’s built-in audio after modifying speak.c with the path of your libaudio.h (e.g., /usr/demo/SOUND/libaudio.h). It’s a simple phoneme concatenation system with commensurate synthesized speech quality (a directory of phoneme audio files is included). [Joe Campbell, jpcampb@afterlife.ncsc.mil]

A public domain version of the same Naval Research Lab text to phoneme rules can be obtained from:

The comp.speech FTP site includes a speech synthesis directory at ftp://svr-ftp.eng.cam.ac.uk/pub/comp.speech/synthesis. The main package is “rsynth” which is a complete text to speech synthesis system. Several component packages are also present. “textnorm” converts non-words such as digit strings into words (e.g. 1000 to ONE THOUSAND). “english2phoneme” does some of the same but its main functionality is to guess an appropriate phoneme sequence for each word. “klatt” takes a parametric form that describes each phoneme and converts it to a waveform. Other packages exist in the same directory to edit and visualise the klatt parameters. [Tony Robinson, ajr@softsound.com]

Q1.3.6: Filter Design Software

Updated Sep 9 2004

There are many filter design programs available via anonymous FTP or by HTTP. The following are summarized here and discussed in greater detail below:

Brian Evans’ and Niranjan Damera-Venkata’s packages for Matlab and Mathematica.

ScopeFIR.

FilterExpress.

Charles Poynton’s filter design resource page.

Juhana Kouhia’s hotlist.

Alex Matulich’s recipes for compiling 2-pole digital filters.

The August 92 issue of IEEE Transactions on Signal Processing includes a paper entitled “METEOR: A Constraint-Based FIR Filter Design Program” by Kenneth Steiglitz, Thomas W. Parks and James F. Kaiser. The authors describe an FIR design program which allows specification of the target frequency response characteristics in a fairly generalized and flexible way. As well as designing filters, the program can optimize filter lengths and push band limits.

The source for the programs (meteor.p, form.p, meteor.c, and form.c) and the METEOR paper as a postscript file may be found at http://www. music.Princeton.edu/classes/class.html. The programs were originally written in Pascal and then evidentally run through p2c to produce the C versions; all the necessary Pascal library stuff is included in the C code and they built error-free out of the box for me on an SGI machine.

There is no manual. The paper includes instructions on running the programs. [Steve Clift, clift@mail.anacapa.net]

Other free filter design packages are DFiltFIR and DFiltInt. DFiltFIR designs minimax approximation FIR filters. It uses the algorithm developed by McClelland and Parks and incorporates constraints on the response as proposed by Grenez. DFiltInt designs minimum mean-square error FIR interpolating filters. The design specification is in terms of a tabulated power spectrum model for the input signal.

Another source is netlib: “A free program to design IIR Butterworth, Chebyshev, and Cauer (elliptic) filters, in any of lowpass, bandpass, band reject, and high pass configurations, is available in netlib (e.g., netlib.bell-labs.com) as the file netlib/cephes/ellf.shar.Z. By email to netlib@netlib.bell-labs.com the request message text is `send ellf from cephes’. The URL is http://www.netlib.org. [Stephen Moshier, moshier@world.std.com]

This program was created and tested using Borland C++ 2.0. This requires a pretty reasonable C++ compiler – it is reported that QuickC (not C++) won’t do it. [Witold Waldman, from Charles Owen at mgcbo@uxa.ecn.bgu.au; also Andrew Ukrainec, ukrainec@InfoUkes.com]

I have developed a MATLAB (vers 4.0 for Windows) program that allows for the frequency domain design of the “near optimal” Kaiser-Bessel window. The program is based upon the three closed form equations developed by Kaiser and Schafer in 1981 that allow for the specification of the time domain window length, and the frequency domain mainlobe width and relative sidelobe amplitude. For signal processing applications where the spectral content of the windowing function is critical so as not to mask adjacent spectra such as radar signal processing applications where a weak target return adjacent to a strong target return could be easily masked by a windowing function that resolves poorly in frequency; this program allows complete frequency domain specification of the spectral characteristics of the windowing function. The current version of this program allows for the user to specify the two frequency domain parameters of mainlobe width and relative sidelobe amplitude and lets the window length fall out as the dependent variable. The program is easily modified to allow for any two parameters to be selected and allowing the third to be determined as a result.

This program will output to an ASCII file the window coefficients that can be easily dumped to an EPROM or included in a program. It also generates both time and frequency domain graphs so that the user can visually verify the widow record length and spectral content. I will gladly provide any interested parties with my MATLAB code.

We have released a set of Matlab packages to optimize the following characteristics of analog filter designs simultaneously:

magnitude response

linear phase in the passband

peak overshoot in the step response

quality factors (Q)

subject to constraints on the same characteristics. The Matlab packages take about 10 seconds for fourth-order filters and 3 minutes for eighth-order filters to run on a 167-MHz Sun Ultra-2 workstation.

We use the symbolic mathematics environment Mathematica to describe the constrained non-linear optimization problem formally, derive the gradients of the cost function and constraints, and synthesize the Matlab code to perform the optimization. In the public release, we provide the Matlab to optimize analog IIR filters of fourth, sixth, and eighth orders. Using the Mathematica formulation, designers can add new measures and constraints, such as capacitance spread for integrated circuit layout, and regenerate the Matlab code.

We describe the framework in [1]. An earlier version of the framework is described in [2]. We plan to extend this framework to digital IIR filters.

ScopeFIR is a FIR filter design tool for Windows 95/NT which designs complex FIR filters using the Parks-McClellan algorithm or windowing. It can then mix, scale, quantize, and edit the FIR coefficients. It creates a wide variety of impulse and frequency response plots, and supports many data file formats, including TI assembly and ADI PM. Shareware with a 60-day trial period, available from http://www.iowegian.com/scopefir.htm.

[Grant Griffin, grant.griffin@iowegian.com]

FilterExpress is a free filter synthesis tool for Windows. It supports the design and analysis of IIR, FIR and multirate FIR filters. It is available for download from http://www.systolix.co.uk/swdownload.htm.

Alex Matulich has compiled recipes (step by step instructions) for coding three kinds of 2-pole digital filters, both low-pass and high-pass, complete with correction factors to ensure that the 3 dB cutoff frequency stays where you put it when you cascade filters of the same type together.

The recipes cover Butterworth, Critically-Damped, and Bessel filters. Alex also includes test results; i.e., plots of actual frequency response and step-function temporal response for each filter.

Q1.3.7: Audio effects

Updated 2/11/02

Harmony Central

Harmony Central publishes some of the source code for its synthesis and audio processing program at http://www.harmony-central.com/Computer/Programming/. The code may be used in public releases, but Harmony Central asks you to credit the author and possibly make the product available for free or publish any modified code.

Music-DSP Source Code Archive

Musicdsp.org is a collection of data gathered for the music dsp community. It includes code for wavetable synthesis, dithering, guitar feedback, and many other effects and algorithms.